Abstract

This paper presents the issue of the innovation policy of real

time audio transmission process in CELT codec using GNU

radio, a SDR by USRP2. As process technology evolves,

processors become more computationally capable pushing

the borderline between software and hardware closer to the

antenna. To serve the purposes of innovation LPC is

necessary for speech coding. The basic foundation of speech

coding is to represent the speech signal with the fewest

number of bits, while maintaining a sufficient level of

quality of the retrieved or synthesized speech with reasonable computational complexity. To achieve high

quality speech at a low bit rate, coding algorithms apply

sophisticated methods to reduce the redundancies, that is, to

remove the irrelevant information from the speech signal.

The paper presents a comprehensive assessment of the

innovation process that, audio transmission using CELP is a

good codec, but for batter performance in real time audio

transmission CELT is the one of the best codec cause of its

codebook and other resources. In this innovation CELT is

the perfect combination with GNU radio for this purposes.

Keyboard : GNU Radio, LPC,CELP, CELT, SDR, USRP2, GMSK, Sampling rate,

Introduction

Most of the wireless system research uses the simulation as

an important tool to validate the system performance. The

motivation of this topics is to build a flexible test bed for

evaluating the novel algorithms under wireless transmission

environment. The rapid growth in wireless communication

systems demands a technology that is capable of conveying

data at high speed and with reliability. The future of

communication is wireless, therefore both research and

testing focus on improvement of the techniques of wireless

transmission. In GNU radio GNU Radio Companion (GRC)

is a graphical tool for creating signal flow graphs and generating flow-graph source code. It is an open-source

Visual programming language for signal processing using

the GNU Radio libraries. The GNU Radio package is

provided with a complete HDTV transmitter and receiver, a

spectrum analyzer, an oscilloscope, a multichannel receiver

and a wide collection of modulators and demodulators. The

user interface is called GNU Radio companion or GRC.

GNU Radio has several blocks that can generate data or

read/write from/to in different formats, like binary complex

values or WAV-files. This Graphical User Interface (GUI)

can be used to recreate any model based on the need. The

GRC helps to easily connect the different modules without

the need of using the command line interface or directly

writing the python codes. GMSK (Gaussian Minimum Shift Keying) Modulation is a

modulation technique that can provide large data rates with

sufficient robustness to radio channel impairments.

GMSK had also achieved popularity for use in commercial high speed broadband wireless systems as the spectrum is

utilized more efficiently. GMSK is now being widely

implemented in high-speed digital communications. GMSK

has been accepted as standard in several wire line and

wireless applications. Thus GMSK is the next generation

transport technology for wireless communications. GMSK is

a special form of multicarrier modulation technique in

which the available bandwidth is divided into many narrow

sub carriers or sub-channels. This allows many users to

transmit in an allocated band in an GMSK system. Each user

is allocated several carriers in which to transmit their data.

The separation of the sub-carriers is such that there is a very compact spectral utilization. With GMSK, it is possible to

have no overlapping sub channels in the frequency domain,

thus increasing the transmission rate. ( Gina Colangelo) [1]

With a SCA (Software Communications Architecture)

implementation like GNU radio project has emerged as one

of the most exciting and promising technology. The GNU

radio system provides an open source software platform

which together with low cost hardware called USRP

(Universal Software Radio Peripheral) can be used to

develop various software radio applications and implement

new technologies for testing purpose. The Software Defined Radio (SDR) allows to bring the code as close to the antenna

as possible and because of this it becomes more convenient

to be used for academic purposes. The code were generated

using C++, Python and XML, which the includes the

processes involved in formation of the GMSK signal for

transmission in modulation techniques and also

demodulation techniques for the received signal. Here is a

technique where more than one sub carriers are used to

transmit a single data. (Mutsawashe Gahadza , Minseok

Kim, Jun-Ichi Takada,)[2]

Real Time Audio Transmission in CELT using GNU Radio By USRP2

Author1: Mohammad Mosfiqur Rahman

Instructor, Department of Computer Technology

Barisal Polytechnic Institute, Barisal, Bangladesh

Author2: Dr Miftahur Rahman

Professor, Department of Mathematics and Physics

North South University, Dhaka, Bangladesh

International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) © International Research Publication House http://www.irphouse.com

445

Figure 1 :Communication model with USRP

Implementing the GMSK signal and thus a better signal can

be retrieved which would otherwise had been distorted and

ultimately lost. Transmitters of this type use GMSK modulation and digital encoding to guarantee protection of

transmitted data. Only special receiver, equipped with

relevant decoder, can receive signals from such transmitters.

Any other receiver provides ―white noise‖ reception only.

GMSK is a simple yet effective approach to digital

modulation for wireless data transmission. GMSK has been

adopted by many wireless data communication protocols.

Key advantages include spectral efficiency, low phase

distortion and coherence of the signal, it also improves noise

immunity when demodulating.

Methods

In most modern paper in the field of audio transmission the

methodological concepts competitively as a set of rules and

practice followed by codec. CELT (Constrained Energy

Lapped Transform) is an open, royalty-free audio

compression format and a free software codec for use in

low-latency audio communication. It is a lossy codec, utput

is split in bands approximating the critical bands; . (Dr.

Jean-Marc Valin,Gregory Maxwell, and Dr. Timothy B. Terriberry)[3]

Sampling rates from 32 kHz to 48 kHz and above can be use

in CELT, adaptive bit-rate from 32 kbit/s to 128 kbit/s per

channel and above. It uses ultra-low algorithmic delay (as

low as 2 ms; scalable, typically from 3 to 9 ms).

One of the very low delay audio codec CELT designed for

high-quality communications. Traditional full-bandwidth

codecs such as Vorbis and AAC can offer high quality but

they require codec delays of hundreds of milliseconds,

which makes them unsuitable for real-time interactive

applications like tele-conferencing. Speech targeted codecs,

such as Speex or G.722, have lower 20-40ms delays but their speech focus and limited sampling rates restricts their

quality, especially for music. Additionally, the other

mandatory components of a full network audio system—

audio interfaces, routers, jitter buffers— each add their own

delay. For lower speed networks the time it takes to serialize

a packet onto the network cable takes considerable time,

and over the long distances the speed of light imposes a

significant delay. In teleconferencing— it is important to

keep delay low so that the participants can communicate

fluidly without talking on top of each other and so that their

own voices don't return after a round trip as an annoying echo. indeed a challenging area in audio codec design,

because as a codec is forced to work on the smaller chunks

of audio required for low delay it has access to less

redundancy and less perceptual information which it can use

to reduce the size of the transmitted audio. CELT is

designed to bridge the gap between "music" and "speech"

codecs, permitting new very high quality teleconferencing

applications, and to go further, permitting latencies much

lower than speech codecs normally provide to enable

applications such as remote musical collaboration even over

long distances. In keeping with the Xiph.Org mission—

CELT is also designed to accomplish this without copyright

or patent encumbrance. Only by keeping the formats that

drive our Internet communication free and unencumbered

can we maximize innovation, collaboration, and interoperability. . There is also a basic tool for testing the

encoder and decoder called68 "testcelt" located in libcelt/:

testcelt <rate> <channels> <frame size> <bytes per packet>

input.sw output.sw, where input.sw is a 16-bit (machine

endian) audio file sampled at 32000 Hz to 96000 Hz. The

output file is already decompressed. For example, for a 44.1

kHz mono stream at ~64kbit/sec and with 256 sample

frames: testcelt 44100 1 256 46 intput.sw output.sw Since

44100/256*46*8 = 63393.74 bits/sec.All even frame sizes

from 64 to 512 are currently supported, although power-of-

two sizes are recommended and most CELT development is

done84 using a size of 256. The delay imposed by CELT is 1.25x - 1.5x the frame duration depending on the frame size

and some details of CELT's internal operation. For 256

sample frames the delay is 1.5x or 384 samples, so the total

codec delay in the above example is 8.70ms

(1000/(44100/384)).

CELT is already ahead of the competition. Its delay:

Configurable, 1.3 ms to 24 ms, ~8 ms typical and quality (at

equivalent rates): Much better than G.722.1C, as good as or

better than AAC-LD, better than ULD. Its flexibility: 24

kbps to 160+ kbps, 32 kHz to 96 kHz, configurable delay,

low-complexity mode The freedom: Open source (BSD), no patents and the transform codec (MDCT, like MP3, Vorbis)

Explicitly code energy of each band of the signal has coarse

shape of sound preserved no matter what and code

remaining details using vector quantization. Also uses pitch

prediction with a time offset, CELT is similar to linear

prediction used by speech codecs and helps compensate for

poor frequency resolution

CELT is short block transform that only capable of

resolving harmonics if the period is an exact multiple of the

frame size. For any other period length, the current window

will contain a portion of the period offset by some phase.

We search the recently decoded signal data for a window that covers the same portion of the period with the same

phase offset. While the harmonics will still not resolve into

distinct MDCT bins, for periodic inputs the predictor will

produce the same pattern of energy spreading. The pitch

predictor is specified by a period defined in the time domain

and a set of gains defined in the frequency domain. The

pitch period is the time offset to the window in the recent

synthesis signal history that best matches the current

encoding window. We estimate the period using the

frequency domain generalized cross-correlation between the

zero-padded input window and the last Lp = 1024 decoded samples .

Two of the parameter sets transmitted to the decoder are

encoded at variable rate: the energy in each band, which is

entropy coded, and the pitch period, which is not transmitted

if the pitch gains are all zero. To achieve a constant bit-rate

without a bit reservoir, we must adapt the rate of the

innovation quantization. We first assume that both the

encoder and the decoder know how many 8-bit bytes are

PC1 USRP1 Air PC2 USRP2

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446

used to encode the frame. This number is either agreed on

when establishing the communication or obtained during the

communication, e.g. the decoder knows the size of any UDP

datagram it receives. Given that, both the encoder and the

decoder can implement the same mechanism to determine

the innovation bit allocation. This mechanism is based solely on the number of bits remaining after encoding the

energy and pitch parameters. A static table determines the

bit-allocation in each band given only the number of bits

available for quantizing the innovation. The correspondence

between the number of bits in a band and the number of

pulses is given by the [6]. For a given number of innovation

bits, the distribution across the bands is constant in time.

This is equivalent to using a psychoacoustic masking.

Each band's share of available bits is fixed, specially CELT

transmits no side information for allocation and it equivalent

to modeling within band masking. The signal-to-mask ratio

for each band is roughly constant, so ignores inter-band masking and tone vs. noise effects.

In this communication one pc with Linux Ubuntu have

proper setup of GNU Radio in tx side and rx side both.

These pc are connected with USRP2 N210 . The

communication media was wireless communication with 2.4

GHz frequency.

Figure2: Basic Architecture of CELT implemented in Linux platform

In present it is become truth that the USRP-2 (Daughter

Board XCVR 2450) is capable to transmit & receive real

time audio signal. It becomes true when in GRC the NBFM

modulation is used in tx & rx end.

Using narrow band frequency modulation the transmission

side of the GRC blocks Audio source, Rational Resampler,

NBFM transmit, UHD:UHD Sink are used. There the

sample rate of audio source is 48KHz and the sample rate of

UHD:UHD Sink is 195.312KHz. And in receiving side the

GRC blocks UHD:UHD Source, NBFM Receiver, WX GUI Scope Sink, Rational Resampler, Multiply Constant, Delay,

Audio Sink are used. Here sample rate of the UHD: UHD

Source is 195.312 KHz and sample rate of audio sink is

48KHz.The output of this transmission & receiver’s quality

was so poor in quality. Very much noise is in the output

signal. Delay, echo is in the signal, so next choice was

WBFM.

Using wide band frequency modulation the transmission

side of the GRC blocks are Audio source, WX GUI Scope

Sink ,Rational Resampler, WBFM transmit, UHD:UHD

Sink. There the sample rate of audio source is 48KHz and

the sample rate of UHD:UHD Sink is 195.312KHz.In receiving side the GRC blocks UHD:UHD Source, WBFM

Receiver, WX GUI Scope Sink, Rational Resampler,

Multiply Constant, Delay, Audio Sink are used. Here sample

rate of the UHD: UHD Source is 195.312 KHz and sample

rate of audio sink is 48KHz.The output of this transmission

& receiver’s quality is still poor in quality. Still noise is in

the output signal, delay, echo is in the signal, so next choice

was GMSK. Using Gaussian Modulation Shift Keying the transmission

side of the GRC blocks are TCP source, Packet Encoder,

GMSK Mod, UHD:UHD Sink . There the sample rate of

packet encoder is 512KHz and the sample rate of

UHD:UHD Sink is 195.312KHz . Address of TCP Source is

127.0.0.1.In receiving side the GRC blocks UHD:UHD

Source, GMSK Demod , WX GUI Scope Sink, Packet

Decoder, TCP Sink are used. Here sample rate of the UHD:

UHD Source is 195.312 KHz and address of TCP sink is

127.0.0.1.The output of this transmission & receiver’s

quality is not so good quality. Still noise is in the output

signal, delay, echo are lightly in the signal. These are happened for miss match of GNU radio internal code which

automatic generated when circuit is design in the GRC and

our code . So there is a opportunity to make it better by

using code directly.

Figure3 : Implemented Structure of GNU Radio

For analysis of CELT performance should have some tasks

for

1. Measure the highest energy with respect to

normalized energy

2. Measure the fft of .wav file we get in output.

3. Divide others quality respect to the highest quality.

4. Find out the best quality energy.

5. Energy change at the respect of frequency change.

There are two lossy audio codecs that are being used

presently. These are classified in two broad classes.:

1. Low delay (15-30 ms) speech codecs that include

G.72x,GSM,AMR,Speex with low sampling rate of 8 KHz

to 16 KHz with limited fidelity. These codecs do not support

music due to low sampling rate. Low delay is a critical

factor for live interaction due to low collision rate during

conversation and reduced echo cancellation. Low delay

codecs are suitable for live music synchronization that

requires delay of less than 25ms.[5]

Python

Code(Tx)

ALSA (Advance

Linux Sound

Architecture)

CELT

Encoder

USRP

(GNU Radio)

ALSA (Advance

Linux Sound

Architecture)

CELT

Decoder

Python

Code(Rx)

USRP

(GNU Radio)

International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) © International Research Publication House http://www.irphouse.com

447

2. General purpose codec’s that include MP3, AAC, Vorbis

with high delay (>100ms) and high sampling rate of 44.1

KHz and higher. These codec’s support CD quality music

with higher fidelity.

Therefore in summarization the two above mentioned codec

categories, one may observe the following advantages and disadvantages:

i. G.722.1C (ITU-T) with 40 ms delay and upto 32 KHz

sampling frequency

ii. AAC-LD (MPEG) with 20-50ms delay and sampling

frequency up to 48KHz.

iii. ULD (Franhofer) with delay less than 10ms and

sampling frequency up to 48 KHz.

iv. On the other hand CELT is an open source codec with a

lot of potential to be competitive with respect to the existing

codec. It has configurable delay in the range of 1.3 ms to 24

ms with much better quality than G.722.1C, AAC-LD, and

ULD. The data rate range from 24kbps to 160kbps and higher.

Result

In this part the transmitted signal is being transformed using

FFT to get the magnitude. Though FFT is in different point

so magnitude will act as an energy. Here the audible

frequency have much more energy as can be seen from Fig 4

to Fig 8The sampling rate of the original signal is 44.KHz,

then it is verified with 8 KHz, 16 KHz, 32 KHz and 41 KHz. The result of energy versus frequency as functions of

sampling frequencies of 8 KHz,16 KHz, 32 KHz,41 KHz

and 44.1 KHz are shown in Fig 4 to Fig 8 respectively.

when we find out these energy variations with different

sampling rate then we find out the normalized energy over

the sampling rates. The normalized energy variation over

sampling rate in GMSK is shown in Figure 9

Figure 4: Energy variation with sampling rate=8 KHz

Figure 5: Energy variation with sampling rate= 16 KHz

Figure 6: Energy variation with sampling rate 32 KHz

Figure 7: Energy variation with sampling rate 41KHz

Figure 8: Energy variation with sampling rate=44.1KHz

Figure 9 Normalized energy versus sampling rate

In GMSK data always transmit from one side to another.

When the modulation period come then there will some

small chunk. The number of byte in every chunk is packet

size. When a packet size increases then GMSK packet

encoding delay will also increases. When packet size is

Normalized Energy versus Sampling Rate

0

0.2

0.4

0.6

0.8

1

1.2

8 16 32 41 44.1

Sampling Rate (kHz)

No

rma

lize

d E

ne

rgy

Normalized Energy

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448

small then in GMSK there is a calculation in between header

and others tasks of modulation for every part. So

throughput become decrease. Here it has been taken a

different number of packet size like : 26bytes, 28 bytes, 30

bytes, 32 bytes, 34 bytes, 36 bytes 38 bytes,40 bytes, 42

bytes, 44 bytes, 46 bytes, 48 bytes, 50 bytes. Then we transmit the data and receive it from receiver. The receiver

pc shows the maximum throughput when it received data

after 5 second average in each calculation showed in Table 1

Table 1: GMSK maximum throughput Packet size in Bytes Maximum

throughput in

kilobit

Maximum

throughput in

kilobyte

26 51.63 6.45

28 55.61 6.95

30 44.67 5.58

32 47.65 5.95

34 50.63 6.32

36 53.61 6.70

38 56.58 7.07

40 59.56 7.44

42 62.54 7.81

44 65.52 8.19

46 54.84 6.85

48 57.23 4.15

50 59.61 7.45

After getting the maximum GMSK throughput in this

variation of different packet size, we make a chart. In Fig 10

there we get that the upward position of GMSK maximum

throughput when packet size is 44 bytes.

Figure 10: GMSK Maximum Throughput

In Table 2 and also in Fig 11 Bit rate (kbps) versus

sampling rate (kHz) is shown with constant packet size( 42

bytes) and sample rate is changing to find out the necessary

throughput and measure the specific sample rate getting in

real time audio transmission in GNU Radio. In this case delay is not available if there is under flow, because when

required bit rate is less then achieved bit rate then underflow

happen. Achieve bit rate is not flat in a certain point because

the maximum achieved bitrates will be in a maximum

sampling rate and it is the maximum speed for transmission

using GNU radio for a particular packet size

Table 2: Throughput of audio Tx & Rx chain of various

sample rate Packet

size

constant

(Bytes)

Sample

Rate

Required

Bit rate

Achieved

Bitrate

for 5 sec

average

(kBps)

Delay Underflow

42

36k 6.04 5.77 No Yes

38k 6.32 6.09 No Yes

40k 6.71 6.41 No Yrs

41k 6.88 6.57 No Yes

41.5k 6.96 6.64 Yes No

42k 7.04 6.73 Yes No

44k 7.38 7.05 Yrs No

46k 7.72 7.37 Yes No

48k 7.99 7.69 Yes No

.

Figure 11: Bit rate vs sampling rate in audio throughput of tx & rx

In Table 3 and in Fig 12 sampling rate is 40 kHz with 256

sample per frame and different size of packet achieved

bitrates is nearly similar as well as the required bitrates. For

audio throughput sample per frame 256 is a optimum point of compression and delay. From this table we found here

that how many data encoded by 256 sample . If the number

of byte is increases then quality become well, but we have to

transmit more data in network transmission. Here it need

such fixed byte per frame which can keep sufficient level of

quality and performance.

Table 3: Throughput of audio Tx & Rx chain of various

packet size: Sampling

Rate kHz

Samples

per

frame

Packet

size

(Byte)

Required

bit rate

(kBps)

Achived

bit rate

for 5 sec

average

(kBps)

Delay Under

flow

40 256

32 5.12 5.13 No Yes

36 5.76 5.77 No Yes

38 6.08 6.08 No Yes

40 6.40 6.41 No Yes

42 6.72 6.72 No Yes

44 7.04 7.05 No Yes

46 7.36 7.37 Yes No

48 7.69 7.69 Yes No

GMSK Maximum Throughput

0

1

2

3

4

5

6

7

8

9

26 28 30 32 34 36 38 40 42 44 46 48 50

Packet Size (Bytes)

Th

rou

gh

pu

t (kB

ps)

Maximum Throughput

Bit Rate versus Sampling Rate in Audio Throughput of Tx &

Rx

0

1

2

3

4

5

6

7

8

9

36 38 40 41 41.5 42 44 46 48

Sampling Rate (kHz)

Bit

Ra

te (

kBp

s)

Required Bit Rate

Achived Bit Rate

International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) © International Research Publication House http://www.irphouse.com

449

Figure 12: Bit rate vs packet size in audio throughput of tx & rx

In Table 4 the limit for sample per packet of CELT is 64 to

512. If the number of sample per packet increases to 512

then the encoding delay will increase. Here encoding delay

means the time to encode a packet means a sound file

transmit in the CELT and come back

Table 4: Constant sample rate vs various frame size

Sample per

frame

Frame size

(Byte)

Performance

256

8 Bad

16 Poor

32 Fair

64 Nice

128 Good

256 Best

In Table 5 the fixed sampling rate in different frame size

with different packet size the delay is not available when

there is underflow. when packet size increasing the delay is

coming and underflow has decreasing. If delay is present

then echo also present and if delay is absent then echo is

also absent.

Table 5: Constant sample rate in different frame size in

various packet size

Sampling Rate

Frame size

Packet size

(Byte)

Delay Under flow

Echo

41000

256

16 No Yes No

40 No Yes No

42 No Yes No

44 No Yes No

46 Yes No No

50 Yes No Yes

300 44 No No No

48 No Yes No

Discussion

CELT exploits the fact that the ear is mainly sensitive to the

amount of energy in each critical band. The MDCT

spectrum is thus divided into 20 bands of roughly one

critical band each, although the lower frequency bands are

wider due to the low MDCT resolution. We refer to these

bands as the energy bands

CELT is still in an early state of development. At this point, two ways of getting involved are: helping design the

algorithm (requires strong DSP knowledge) or building

applications using CELT. Our feedback can help define the

future direction the codec will take. It applies some of the

CELP principles, but does everything in the frequency

domain, which removes some of the limitations of CELP.

CELT is suitable for both speech and music [4] There are two program languages used in GNU Radio, C++

and Python which play different roles in the whole system.

All the signal processing and performance-critical blocks are

written in C++. Python is used to create a network or graph

and glue these blocks together.

Conclusion

The real time audio transmission using CELT was done

successfully using GNU radio system with its required

hardware and software components. The result shows us the significant performance of the model used to transmit audio

transmission in CELT. From the results and analysis it is

quite clear that the GNU Radio is usable for real life audio

transmission model since the results showed that the original

signal could be retrieved almost without noises. The model

was tested in 2.4 GHz with CELT. CELT brings CD-quality

sound to VoIP-style low-delay applications and better than

MP3 and <10 ms delay.

Future work in this area might look into the application of

GNU Radio in consumer facing applications. Exploring

extremely low latency transport-layer-adaptive synchronized audio-video transmission might also be of interest. Aspects

of CELT that can be improved include dynamic rate

allocation, stereo coupling and pitch prediction. The

transport layer software can also be tested at other center

frequencies such as 5GHz. Some emerging uses of software

defined radio are 4G LTE, WiMax, WiFi, Digital TV,

HDTV, mobile TV etc. The application of low latency

codecs such as CELT in these areas could lead to

groundbreaking results. The combined application of

exceptional codecs such as CELT and adaptive

communication systems such as GNU Radio can

revolutionize the communications sector.

References

I. Gina Colangelo ―Introduction to GMSK:

Gaussian Filtered Minimum Shift Keying‖

(EE194 – SDR)

II. Mutsawashe Gahadza , Minseok Kim, Jun-Ichi

Takada, ―Implementation of a Channel Sounder

using GNU Radio Opensource SDR Platform‖,

Graduate School of Engineering, Tokyo Institute of Technology, 2009.

III. Dr. Jean-Marc Valin,Gregory Maxwell, and Dr.

Timothy B. Terriberry ―CELT: A Low-latency,

High-quality Audio Codec “, 2010

IV. Jean-Marc Valin, Member, IEEE, Timothy B.

Terriberry, Christopher Montgomery, Gregory

Maxwell ― A High-Quality Speech and Audio

Codec With Less than 10 ms Delay” 2010

V. Jean-Marc Valin, Timothy B. Terriberry, Gregory

Maxwell ―A FULL-BANDWIDTH AUDIO

CODECWITH LOW COMPLEXITY AND

VERY LOW DELAY” 2010 VI. www.gnuradio.org, February, 2011

Bit rate versus Packet Size in Audio Throughput of Tx & Rx

0

1

2

3

4

5

6

7

8

9

32 36 38 40 42 44 46 48

Packet Size (Byte)

Bit R

ate

(kB

ps)

Required Bit Rate

Achive Bit Rate

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